Ken,
Sign me up as a beta tester. If you don't want beta testers then sign
me up as a scumbag leech who wants to check out your cool software
without paying anything! :D
Seriously, I'm available if you'd like to get some alternate viewpoints
on how your stuff works.
-MC
___
Thanks a lot. I intend to use it mostly as a SIP user directory. For the
dial-plan I dont mind parsing and syncing XML file across servers (if
there were a small cluster). The main deal is AUTHENTICATION. The
authentication scheme I wish to keep is Kerberos (with SASL in Ldap for
binding). This way
a sip 302 response is Moved Temporarily. The sip stack automatically
follows the 302 to the next location, so you can not do your own
actions based on those. Terminal sip response codes like 486 get
translated to their companion q.850 codes. You can adjust routing
based on other response
I've been looking around a little bit and I haven't seen anything on this yet
so I wanted to ask here.
I want to have a dialplan that captures certain SIP responses and takes
different actions based on the response received.
For example, if I place a call out to a peer and get back "486 Busy he
At one point I was very interested in this...then I got a job. =[
I thought mod_ldap was more of a PoC than anything. It might work (I
couldn't get it working and unfortunately don't remember exactly why..)
but there really isn't much point. I would have to do at least 5 ldap
queries (if not m
Brian West wrote:
> What else is great is his phone does G722 wideband. w00t!
>
btw, what kind of G.722 wideband codec do you support?
I see that you are using G.722 implementation from the "voipcodecs"
library, and that you are using different samplerates, one signalled
in SDP (8000 Hz) and 16
Yes that is correct.. you say 8000 but its really 16000 anyone that
says 16000 is doing it wrong.
/b
On May 28, 2008, at 1:01 PM, Alfred E. Heggestad wrote:
> Brian West wrote:
>> What else is great is his phone does G722 wideband. w00t!
>
> btw, what kind of G.722 wideband codec do you suppo
i'work on a call center with 50 seats, i'can test any dialer based on
FS if you want.
On Wed, May 28, 2008 at 2:22 PM, Ken Rice <[EMAIL PROTECTED]> wrote:
> We're actually in the process of building a new dialer based on
> freeswitch... I don't see how people can actually deal with vicidial... I'v
We¹re actually in the process of building a new dialer based on
freeswitch... I don¹t see how people can actually deal with vicidial... I¹ve
used it and Iw as horrified by the meetme requirements and all kinds of
other fun stuff that just tremondously added to the the load... Not to
mention lack of
if can run vicidial under FS that would be owesome !!!
On Wed, May 28, 2008 at 1:52 PM, Michael Collins <[EMAIL PROTECTED]> wrote:
> Awesome. Now if we could just get astgui/vicidial guys to port their stuff
> over… :D
>
>
>
> -MC
>
>
>
>
>
> From: [EMAIL PROTECTE
Heh. Maybe you could send me what you're using so I don't have to
reinvent the wheel all by myself! :P
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ken
Rice
Sent: Wednesday, May 28, 2008 9:50 AM
To: freeswitch-users@lists.freeswitch.org
Subje
Oh hell no
From: Michael Collins <[EMAIL PROTECTED]>
Reply-To:
Date: Wed, 28 May 2008 09:52:21 -0700
To:
Subject: Re: [Freeswitch-users] RESTful Bounty . Putting the
competitionto REST with RestFul VoIP2.x services
Awesome. Now if we could just get astgui/vicidial guys to port their stuf
Awesome. Now if we could just get astgui/vicidial guys to port their
stuff over... :D
-MC
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ward
Mundy
Sent: Wednesday, May 28, 2008 5:33 AM
To: freeswitch-users@lists.freeswitch.org
Subject: Re:
We have a concept called the "directory interface" not to be confused with
the "user directory".
The directory interface is a pluggable abstract API that looks and feels
like LDAP only you can plug in anything you want to implement the functions.
mod_ldap is a module that registers to this interf
Yes,
We have experimental support for 2 ways to do ss7, telco bridges (not in
tree but privately under development)
and also Sangoma SS7Boost using ss7box (in tree in the OpenZAP lib dist with
FreeSWITCH). Both are new and need testing.
We also have plans to add native isup support to OpenZAP as
There is a prototype developed in OpenZAP to allow FreeSWITCH to act in
place of sangoma SMG in your model.
SS7 Links< > FreeSwitch(OpenZAP) <>Broadcasting Application (
SIP based)
OpenZAP has a new protocol plugin that can communicate with the other end of
Sangoma SMG using the same
By default sip registrations persist via the embedded sqlite but there
is an odbc dsn you can configure in the sofia profile as well. This
is true for most places we use embedded sqlite in the core.
Mike
On May 28, 2008, at 8:55 AM, Aadilkhan Maniyar wrote:
Hi All,
Does FreeSwitch suppor
Currently the directory interface is only used for that dialplan, I
would like to enhance that in the future. The directory dialploan
uses a filter of exten=destination number, and then has name/value
pairs, I will see if I can find the schema we used back when we
developed it, short of th
Here is some info on it
http://wiki.freeswitch.org/wiki/Mod_spidermonkey_odbc
-E
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Hi All,
Does FreeSwitch support the MySQL database?
I could not find any documentation related to the same in the FS wiki.
If FS does not support MySQL or unixODBC how are SIP registrations made
persistent?
Regards,
Aadil
___
Freeswitch-users mail
Hi,
I already had a solution from one FS user which is essentially to append an
extension to the URI used for mapping in DIDWW.
Thanks,
Klaus.
Original-Nachricht
> Datum: Wed, 28 May 2008 16:17:44 +0800
> Von: "Pete Kay" <[EMAIL PROTECTED]>
> An: freeswitch-users@lists.freeswi
Hi all,
A colleague of mine who currently uses Asterisk was asking me the other day
about SS7 support in FreeSWITCH - and whether people have such a thing in
production. I don't think he's looking for MAP support, which makes life a
little easier.
I know of at least one company that's used FS wit
> On Tue, May 27, 2008, Michael Collins wrote:
>
>
> Re: Any chance we'll see some of your sweet recipes on nerdvittles with a
little FS baked in? :D
>
You bet. In fact, we're raising a bounty on the trixbox forums to port
FreePBX to FreeSwitch as we speak...
http://trixbox.org/forums/trixbox-fo
need billing for freeswitch [EMAIL PROTECTED]
On 5/27/08, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
>
>
> HI
>
> We want to try generate 5000 simultanious Voice broadcast calls .
> can the below config will work?
>
> SS7 Links< > Sangoma SMG<--->FreeSwitch<>Broadcasting
> Appli
First of all- Amazing project. Tired of asterisk deadlocking all the
time we have been deploying asterisk with OpenSER as the registrar.
Freeswitch is a huge relief!
This is an extremely important feature we have been looking for.
Asterisk realtime ldap integration is very flaky. I found this pag
HI
We want to try generate 5000 simultanious Voice broadcast calls .
can the below config will work?
SS7 Links< > Sangoma SMG<--->FreeSwitch<>Broadcasting
Application ( SIP based)
Thank you
Imthiyaz
mail2web.
http://wiki.freeswitch.org/wiki/Wiki_meet_2008_05_28
Please add anything to this list you think we need to talk about. As
usual we'll breeze thru jira, docs and various other things that come
up along the way. Please remember to DIGG the 1.0 release article at
http://digg.com/software/FreeS
Hi,
Have you tried the acl function in freeswitch. For example, setup the DIDWW
domain as an acl so no need to use username/password from that domain.
In the default profile, you can specify the default context for that
incoming call to go to in the dialplan.
If my solution works for you, please
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